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64 lines
2.8 KiB
Text
64 lines
2.8 KiB
Text
The RTP libraries
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---------------------
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RTP Buffers
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-----------
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The real time protocol as described in RFC 3550 requires the use of special
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packets containing an additional RTP header of at least 12 bytes. GStreamer
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provides some helper functions for creating and parsing these RTP headers.
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The result is a normal #GstBuffer with an additional RTP header.
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RTP buffers are usually created with gst_rtp_buffer_new_allocate() or
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gst_rtp_buffer_new_allocate_len(). These functions create buffers with a
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preallocated space of memory. It will also ensure that enough memory
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is allocated for the RTP header. The first function is used when the payload
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size is known. gst_rtp_buffer_new_allocate_len() should be used when the size
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of the whole RTP buffer (RTP header + payload) is known.
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When receiving RTP buffers from a network, gst_rtp_buffer_new_take_data()
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should be used when the user would like to parse that RTP packet. (TODO Ask
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Wim what the real purpose of this function is as it seems to simply create a
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duplicate GstBuffer with the same data as the previous one). The
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function will create a new RTP buffer with the given data as the whole RTP
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packet. Alternatively, gst_rtp_buffer_new_copy_data() can be used if the user
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wishes to make a copy of the data before using it in the new RTP buffer.
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It is now possible to use all the gst_rtp_buffer_get_*() or
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gst_rtp_buffer_set_*() functions to read or write the different parts of the
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RTP header such as the payload type, the sequence number or the RTP
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timestamp. The use can also retrieve a pointer to the actual RTP payload data
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using the gst_rtp_buffer_get_payload() function.
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RTP Base Payloader Class (GstBaseRTPPayload)
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--------------------------------------------
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All RTP payloader elements (audio or video) should derive from this class.
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RTP Base Audio Payloader Class (GstBaseRTPAudioPayload)
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-------------------------------------------------------
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This base class can be tested through it's children classes. Here is an
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example using the iLBC payloader (frame based).
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For 20ms mode :
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GST_DEBUG="basertpaudiopayload:5" gst-launch-1.0 fakesrc sizetype=2
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sizemax=114 datarate=1900 ! audio/x-iLBC, mode=20 ! rtpilbcpay
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max-ptime="40000000" ! fakesink
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For 30ms mode :
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GST_DEBUG="basertpaudiopayload:5" gst-launch-1.0 fakesrc sizetype=2
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sizemax=150 datarate=1662 ! audio/x-iLBC, mode=30 ! rtpilbcpay
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max-ptime="60000000" ! fakesink
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Here is an example using the uLaw payloader (sample based).
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GST_DEBUG="basertpaudiopayload:5" gst-launch-1.0 fakesrc sizetype=2
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sizemax=150 datarate=8000 ! audio/x-mulaw ! rtppcmupay max-ptime="6000000" !
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fakesink
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RTP Base Depayloader Class (GstBaseRTPDepayload)
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------------------------------------------------
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All RTP depayloader elements (audio or video) should derive from this class.
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